1. Field of the Invention
This invention relates generally to improvements in systems for audio signal processing and, more particularly, to a new and improved system for sample accurate audio state update to insure that audio parameters are modified precisely at specified sample times.
2. Description of the Related Art
It is common practice in the recording/playback arts, i.e., those associated with film dubbing, musical recording and the like, to integrate plural channels of audio for mixing and/or editing the collective data base of audio information so that it is properly inserted in time synchronism with associated scenes on film or video tracks. In this regard, in recording and playback of multiple audio channels, it is absolutely necessary for all channels to remain in synchronization in order to avoid artifacts such as echoes and distortion. Unfortunately, when handling multiple channels of audio simultaneously, "punching in" (directing audio input to be recorded) and "punching out" (directing audio signals to be provided as output) repetitively for both recording and playback, it has become difficult to smoothly fade between events, modify audio parameters, such as gain, rate and the like, and obtain audio data flow which does not include audio "glitches" or artifacts due to mismatch or out of synch data samples. Much of these artifacts are due to individual and accumulated latencies inherent in the various subsystems and various typical operational procedures associated with such film dubbing systems.
Hence, those concerned with the development and use of improved dubbing systems and the like have long recognized the need for improved systems for sample accurate audio updating which avoids artifacts. The present invention clearly fulfills these needs.